Hi! Thank you so much for offering to help, I really appreciate it. I'm currently testing a custom WebSocket bridge via AudioSocket in Asterisk just to keep the integration moving as a temporary workaround. But I definitely want to fix the SIP trunk issue! I will capture a fresh pjsip set logger on trace with the exact INVITE -> 401 -> ACK -> INVITE (auth) -> 403 exchange and send it to you a bit later. Can I DM it to you so I can safely redact my passwords?